December 2013 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [RFC][PATCH v1] ASoC: fsl_ssi: Add DAI master mode support for SSI on i.MX series
by Nicolin Chen 13 Jun '14

13 Jun '14

From: Nicolin Chen <b42378(a)freescale.com> This patch adds three main functions for DAI master mode: set_dai_fmt(), set_dai_sysclk() and set_dai_tdm_slot(), and one essential baud clock accordingly. After appending this patch, the fsl_ssi driver on i.MX series has the ability to derive LRCLK and BCLK from baud clock source so as to support some audio Codecs which can only be used in slave mode. Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- sound/soc/fsl/fsl_ssi.c | 280 +++++++++++++++++++++++++++++++++++++++++++++++- sound/soc/fsl/fsl_ssi.h | 2 + 2 files changed, 278 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index f9f4569..b2ebaf8 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -38,6 +38,7 @@ #include <linux/device.h> #include <linux/delay.h> #include <linux/slab.h> +#include <linux/spinlock.h> #include <linux/of_address.h> #include <linux/of_irq.h> #include <linux/of_platform.h> @@ -139,7 +140,10 @@ struct fsl_ssi_private { bool ssi_on_imx; bool imx_ac97; bool use_dma; + bool baudclk_locked; u8 i2s_mode; + spinlock_t baudclk_lock; + struct clk *baudclk; struct clk *clk; struct snd_dmaengine_dai_dma_data dma_params_tx; struct snd_dmaengine_dai_dma_data dma_params_rx; @@ -434,13 +438,18 @@ static int fsl_ssi_startup(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(rtd->cpu_dai); + unsigned long flags; /* First, we only do fsl_ssi_setup() when SSI is going to be active. * Second, fsl_ssi_setup was already called by ac97_init earlier if * the driver is in ac97 mode. */ - if (!dai->active && !ssi_private->imx_ac97) + if (!dai->active && !ssi_private->imx_ac97) { fsl_ssi_setup(ssi_private); + spin_lock_irqsave(&ssi_private->baudclk_lock, flags); + ssi_private->baudclk_locked = false; + spin_unlock_irqrestore(&ssi_private->baudclk_lock, flags); + } return 0; } @@ -502,6 +511,243 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, } /** + * fsl_ssi_set_dai_fmt - configure Digital Audio Interface Format. + */ +static int fsl_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) +{ + struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai); + struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + u32 strcr = 0, stcr, srcr, scr, mask; + + scr = read_ssi(&ssi->scr) & ~(CCSR_SSI_SCR_SYN | CCSR_SSI_SCR_I2S_MODE_MASK); + scr |= CCSR_SSI_SCR_NET; + + mask = CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TFDIR | CCSR_SSI_STCR_TXDIR | + CCSR_SSI_STCR_TSCKP | CCSR_SSI_STCR_TFSI | CCSR_SSI_STCR_TFSL | + CCSR_SSI_STCR_TEFS; + stcr = read_ssi(&ssi->stcr) & ~mask; + srcr = read_ssi(&ssi->srcr) & ~mask; + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + ssi_private->i2s_mode = CCSR_SSI_SCR_I2S_MODE_MASTER; + break; + case SND_SOC_DAIFMT_CBM_CFM: + ssi_private->i2s_mode = CCSR_SSI_SCR_I2S_MODE_SLAVE; + break; + default: + return -EINVAL; + } + scr |= ssi_private->i2s_mode; + + /* Data on rising edge of bclk, frame low, 1clk before data */ + strcr |= CCSR_SSI_STCR_TFSI | CCSR_SSI_STCR_TSCKP | + CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TEFS; + break; + case SND_SOC_DAIFMT_LEFT_J: + /* Data on rising edge of bclk, frame high */ + strcr |= CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TSCKP; + break; + case SND_SOC_DAIFMT_DSP_A: + /* Data on rising edge of bclk, frame high, 1clk before data */ + strcr |= CCSR_SSI_STCR_TFSL | CCSR_SSI_STCR_TSCKP | + CCSR_SSI_STCR_TXBIT0 | CCSR_SSI_STCR_TEFS; + break; + case SND_SOC_DAIFMT_DSP_B: + /* Data on rising edge of bclk, frame high */ + strcr |= CCSR_SSI_STCR_TFSL | CCSR_SSI_STCR_TSCKP | + CCSR_SSI_STCR_TXBIT0; + break; + default: + return -EINVAL; + } + + /* DAI clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + /* Nothing to do for both normal cases */ + break; + case SND_SOC_DAIFMT_IB_NF: + /* Invert bit clock */ + strcr ^= CCSR_SSI_STCR_TSCKP; + break; + case SND_SOC_DAIFMT_NB_IF: + /* Invert frame clock */ + strcr ^= CCSR_SSI_STCR_TFSI; + break; + case SND_SOC_DAIFMT_IB_IF: + /* Invert both clocks */ + strcr ^= CCSR_SSI_STCR_TSCKP; + strcr ^= CCSR_SSI_STCR_TFSI; + break; + default: + return -EINVAL; + } + + /* DAI clock master masks */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + strcr |= CCSR_SSI_STCR_TFDIR | CCSR_SSI_STCR_TXDIR; + scr |= CCSR_SSI_SCR_SYS_CLK_EN; + break; + case SND_SOC_DAIFMT_CBM_CFM: + scr &= ~CCSR_SSI_SCR_SYS_CLK_EN; + break; + default: + return -EINVAL; + } + + stcr |= strcr; + srcr |= strcr; + + if (ssi_private->cpu_dai_drv.symmetric_rates) { + /* Need to clear RXDIR when using SYNC mode */ + srcr &= ~CCSR_SSI_SRCR_RXDIR; + scr |= CCSR_SSI_SCR_SYN; + } + + write_ssi(stcr, &ssi->stcr); + write_ssi(srcr, &ssi->srcr); + write_ssi(scr, &ssi->scr); + + return 0; +} + +/** + * fsl_ssi_set_dai_sysclk - configure Digital Audio Interface bit clock + * + * Note: This function can be only called when using SSI as DAI master + * + * Quick instruction for parameters: + * freq: Output BCLK frequency = samplerate * 32 (fixed) * channels + * dir: SND_SOC_CLOCK_OUT -> TxBCLK, SND_SOC_CLOCK_IN -> RxBCLK. + */ +static int fsl_ssi_set_dai_sysclk(struct snd_soc_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai); + struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + int synchronous = ssi_private->cpu_dai_drv.symmetric_rates, ret; + u32 pm = 999, div2, psr, stccr, mask, afreq, factor, i; + unsigned long flags, clkrate, baudrate, tmprate; + u64 sub, savesub = 100000; + + /* Don't apply it to any non-baudclk circumstance */ + if (IS_ERR(ssi_private->baudclk)) + return -EINVAL; + + /* It should be already enough to divide clock by setting pm alone */ + psr = 0; + div2 = 0; + + factor = (div2 + 1) * (7 * psr + 1) * 2; + + for (i = 0; i < 255; i++) { + /* The bclk rate must be smaller than 1/5 sysclk rate */ + if (factor * (i + 1) < 5) + continue; + + tmprate = freq * factor * (i + 2); + clkrate = clk_round_rate(ssi_private->baudclk, tmprate); + + do_div(clkrate, factor); + afreq = (u32)clkrate / (i + 1); + + if (freq == afreq) + sub = 0; + else if (freq / afreq == 1) + sub = freq - afreq; + else if (afreq / freq == 1) + sub = afreq - freq; + else + continue; + + /* Calculate the fraction */ + sub *= 100000; + do_div(sub, freq); + + if (sub < savesub) { + baudrate = tmprate; + savesub = sub; + pm = i; + } + + /* We are lucky */ + if (savesub == 0) + break; + } + + /* No proper pm found if it is still remaining the initial value */ + if (pm == 999) { + dev_err(cpu_dai->dev, "failed to handle the required sysclk\n"); + return -EINVAL; + } + + stccr = CCSR_SSI_SxCCR_PM(pm + 1) | (div2 ? CCSR_SSI_SxCCR_DIV2 : 0) | + (psr ? CCSR_SSI_SxCCR_PSR : 0); + mask = CCSR_SSI_SxCCR_PM_MASK | CCSR_SSI_SxCCR_DIV2 | CCSR_SSI_SxCCR_PSR; + + if (dir == SND_SOC_CLOCK_OUT || synchronous) + write_ssi_mask(&ssi->stccr, mask, stccr); + else + write_ssi_mask(&ssi->srccr, mask, stccr); + + spin_lock_irqsave(&ssi_private->baudclk_lock, flags); + if (!ssi_private->baudclk_locked) { + ret = clk_set_rate(ssi_private->baudclk, baudrate); + if (ret) { + spin_unlock_irqrestore(&ssi_private->baudclk_lock, flags); + dev_err(cpu_dai->dev, "failed to set baudclk rate\n"); + return -EINVAL; + } + ssi_private->baudclk_locked = true; + } + spin_unlock_irqrestore(&ssi_private->baudclk_lock, flags); + + return 0; +} + +/** + * fsl_ssi_set_dai_tdm_slot - set TDM slot number + * + * Note: This function can be only called when using SSI as DAI master + */ +static int fsl_ssi_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, u32 tx_mask, + u32 rx_mask, int slots, int slot_width) +{ + struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(cpu_dai); + struct ccsr_ssi __iomem *ssi = ssi_private->ssi; + u32 val; + + /* The slot number should be >= 2 if using Network mode or I2S mode */ + val = read_ssi(&ssi->scr) & (CCSR_SSI_SCR_I2S_MODE_MASK | CCSR_SSI_SCR_NET); + if (val && slots < 2) { + dev_err(cpu_dai->dev, "slot number should be >= 2 in I2S or NET\n"); + return -EINVAL; + } + + write_ssi_mask(&ssi->stccr, CCSR_SSI_SxCCR_DC_MASK, + CCSR_SSI_SxCCR_DC(slots)); + write_ssi_mask(&ssi->srccr, CCSR_SSI_SxCCR_DC_MASK, + CCSR_SSI_SxCCR_DC(slots)); + + /* The register SxMSKs needs SSI to provide essential clock due to + * hardware design. So we here temporarily enable SSI to set them. + */ + val = read_ssi(&ssi->scr) & CCSR_SSI_SCR_SSIEN; + write_ssi_mask(&ssi->scr, 0, CCSR_SSI_SCR_SSIEN); + + write_ssi(tx_mask, &ssi->stmsk); + write_ssi(rx_mask, &ssi->srmsk); + + write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_SSIEN, val); + + return 0; +} + +/** * fsl_ssi_trigger: start and stop the DMA transfer. * * This function is called by ALSA to start, stop, pause, and resume the DMA @@ -517,6 +763,7 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, struct fsl_ssi_private *ssi_private = snd_soc_dai_get_drvdata(rtd->cpu_dai); struct ccsr_ssi __iomem *ssi = ssi_private->ssi; unsigned int sier_bits; + unsigned long flags; /* * Enable only the interrupts and DMA requests @@ -557,8 +804,12 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_RE, 0); if (!ssi_private->imx_ac97 && (read_ssi(&ssi->scr) & - (CCSR_SSI_SCR_TE | CCSR_SSI_SCR_RE)) == 0) + (CCSR_SSI_SCR_TE | CCSR_SSI_SCR_RE)) == 0) { write_ssi_mask(&ssi->scr, CCSR_SSI_SCR_SSIEN, 0); + spin_lock_irqsave(&ssi_private->baudclk_lock, flags); + ssi_private->baudclk_locked = false; + spin_unlock_irqrestore(&ssi_private->baudclk_lock, flags); + } break; default: @@ -585,6 +836,9 @@ static int fsl_ssi_dai_probe(struct snd_soc_dai *dai) static const struct snd_soc_dai_ops fsl_ssi_dai_ops = { .startup = fsl_ssi_startup, .hw_params = fsl_ssi_hw_params, + .set_fmt = fsl_ssi_set_dai_fmt, + .set_sysclk = fsl_ssi_set_dai_sysclk, + .set_tdm_slot = fsl_ssi_set_dai_tdm_slot, .trigger = fsl_ssi_trigger, }; @@ -897,6 +1151,9 @@ static int fsl_ssi_probe(struct platform_device *pdev) /* Older 8610 DTs didn't have the fifo-depth property */ ssi_private->fifo_depth = 8; + ssi_private->baudclk_locked = false; + spin_lock_init(&ssi_private->baudclk_lock); + if (of_device_is_compatible(pdev->dev.of_node, "fsl,imx21-ssi")) { u32 dma_events[2]; ssi_private->ssi_on_imx = true; @@ -914,6 +1171,15 @@ static int fsl_ssi_probe(struct platform_device *pdev) goto error_irqmap; } + /* For those SLAVE implementations, we ingore non-baudclk cases + * and, instead, abandon MASTER mode that needs baud clock. + */ + ssi_private->baudclk = devm_clk_get(&pdev->dev, "baud"); + if (IS_ERR(ssi_private->baudclk)) + dev_warn(&pdev->dev, "could not get baud clock: %d\n", ret); + else + clk_prepare_enable(ssi_private->baudclk); + /* * We have burstsize be "fifo_depth - 2" to match the SSI * watermark setting in fsl_ssi_startup(). @@ -1059,8 +1325,11 @@ error_dev: device_remove_file(&pdev->dev, dev_attr); error_clk: - if (ssi_private->ssi_on_imx) + if (ssi_private->ssi_on_imx) { + if (!IS_ERR(ssi_private->baudclk)) + clk_disable_unprepare(ssi_private->baudclk); clk_disable_unprepare(ssi_private->clk); + } error_irqmap: irq_dispose_mapping(ssi_private->irq); @@ -1076,8 +1345,11 @@ static int fsl_ssi_remove(struct platform_device *pdev) platform_device_unregister(ssi_private->pdev); snd_soc_unregister_component(&pdev->dev); device_remove_file(&pdev->dev, &ssi_private->dev_attr); - if (ssi_private->ssi_on_imx) + if (ssi_private->ssi_on_imx) { + if (!IS_ERR(ssi_private->baudclk)) + clk_disable_unprepare(ssi_private->baudclk); clk_disable_unprepare(ssi_private->clk); + } irq_dispose_mapping(ssi_private->irq); return 0; diff --git a/sound/soc/fsl/fsl_ssi.h b/sound/soc/fsl/fsl_ssi.h index e6b9a69..e6b6324 100644 --- a/sound/soc/fsl/fsl_ssi.h +++ b/sound/soc/fsl/fsl_ssi.h @@ -125,7 +125,9 @@ struct ccsr_ssi { #define CCSR_SSI_SRCR_REFS 0x00000001 /* STCCR and SRCCR */ +#define CCSR_SSI_SxCCR_DIV2_SHIFT 18 #define CCSR_SSI_SxCCR_DIV2 0x00040000 +#define CCSR_SSI_SxCCR_PSR_SHIFT 17 #define CCSR_SSI_SxCCR_PSR 0x00020000 #define CCSR_SSI_SxCCR_WL_SHIFT 13 #define CCSR_SSI_SxCCR_WL_MASK 0x0001E000 -- 1.8.4

4 9

[alsa-devel] Segfaults in snd_pcm_area_silence() when using softvol
by Tanu Kaskinen 18 May '14

18 May '14

Hi, A bug was reported to pulseaudio about a segfault in alsa-lib code when using softvol: https://bugs.freedesktop.org/show_bug.cgi?id=51060 . It has been observed by several people, and this mail was prompted by the last bug comment, which rightly recommends getting in touch with the alsa developers: > I've had this problem too and the problem only occurs with > -ftree-vectorize. > > It looks to me as if this is a gcc bug, as it produces a "movdqa" or > "vmovdqa" instruction which is used with an unaligned pointer. > > Now, I have no clue why gcc assumes the pointer would be an aligned > one, maybe it is even right and something else is wrong, but I haven't > spent too much time digging into this. > > (note that it still happens with gcc 4.7 and the current 4.8 > development one) > > Also, this bug should probably be brought up with the alsa guys as > well, as I don't think pulseaudio is at fault here. Here's the stack trace for your convenience (see the linked bug for more details) #0 snd_pcm_area_silence (dst_area=0x7fffe875bb00, dst_offset=<optimized out>, samples=2, format=SND_PCM_FORMAT_S16_LE) at pcm.c:2549 #1 0x00007fffe87b4398 in snd_pcm_areas_silence (dst_areas=0x5555557fa100, dst_offset=891, channels=2, frames=439, format=SND_PCM_FORMAT_S16_LE) at pcm.c:2662 #2 0x00007fffe87f9bdb in softvol_convert_stereo_vol (svol=0x555555813cf0, dst_areas=0x5555557fa0e0, dst_offset=891, src_areas=0x5555557fa0e0, src_offset=<optimized out>, channels=2, frames=439) at pcm_softvol.c:286 #3 0x00007fffe87fae58 in snd_pcm_softvol_write_areas (pcm=0x55555581db60, areas=0x5555557fa0e0, offset=891, size=439, slave_areas=0x5555557fa0e0, slave_offset=891, slave_sizep=0x7fffe875bcd0) at pcm_softvol.c:607 #4 0x00007fffe87c6fc7 in snd_pcm_plugin_mmap_commit (pcm=0x55555581db60, offset=<optimized out>, size=439) at pcm_plugin.c:424 #5 0x00007fffe87b9676 in snd_pcm_mmap_commit (pcm=<optimized out>, offset=<optimized out>, frames=<optimized out>) at pcm.c:6584 #6 0x00007fffe8a74876 in mmap_write (on_timeout=false, polled=false, sleep_usec=<synthetic pointer>, u=0x5555557e6d40) at modules/alsa/alsa-sink.c:636 #7 thread_func (userdata=0x5555557e6d40) at modules/alsa/alsa-sink.c:1694 #8 0x00007ffff722c039 in internal_thread_func (userdata=0x55555587ef80) at pulsecore/thread-posix.c:83 #9 0x00007ffff4a24e2c in ?? () from /lib64/libpthread.so.0 #10 0x00007ffff3ccc1dd in clone () from /lib64/libc.so.6 -- Tanu

2 1

[alsa-devel] [PATCH v3 0/5] ALSA: hda - hdmi: ATI/AMD multi-channel and HBR support
by Anssi Hannula 14 May '14

14 May '14

Hi all! Here's another revision of the ATI/AMD multichannel+HBR patchset. Changes: - added customizable callbacks instead of polluting generic code - fixed wrong codec id in rev3+ check (as previously noted) - cosmetics I've only tested this on NVIDIA/Intel, but I'd guess at least Peter should be able to test this on AMD in a few days or so. The earlier revisions were tested to fully work. Combined patch can be found at: http://onse.fi/files/atihdmi5.patch However, this patchset is on top recent sound-next commits so it does not directly apply on older kernels. Olivier, I hope you will be able to test this at some point as well (same things as last time), even though it will be a bit more work :) Anssi Hannula (5): ALSA: hda - hdmi: Allow HDA patches to customize more operations ALSA: hda - hdmi: Add ATI/AMD multi-channel audio support ALSA: hda - hdmi: Add ELD emulation for ATI/AMD codecs ALSA: hda - hdmi: Add HBR bitstreaming support for ATI/AMD HDMI codecs ALSA: hda - hdmi: Disable ramp-up/down for non-PCM on AMD codecs sound/pci/hda/hda_eld.c | 151 +++++++++++ sound/pci/hda/hda_local.h | 4 + sound/pci/hda/patch_hdmi.c | 643 +++++++++++++++++++++++++++++++++++++-------- 3 files changed, 689 insertions(+), 109 deletions(-) -- Anssi Hannula

9 70

[alsa-devel] Ambiguous DAPM widget names and DPCM
by Daniel Mack 21 Mar '14

21 Mar '14

Hi Mark, Hi Liam, I spent some hours reverse-engineering the DPCM code, and I plan to prepare a patch to add some documentation once I've assembled a fully working setup. However, I have a general question regarding DAPM widget names. DPCM uses DAPM widgets and routes in order to determine the FE <-> BE connections at runtime. The problem with this is that widget names are very ambiguous, as the automatically created widgets for DAIs are named after their stream_name properties, which contain "Playback" or "Capture" for all components in my case. Hence, the code which walks the connections (snd_soc_dapm_dai_get_connected_widgets()) will most likely match the wrong one (which has empty sound/sink lists), which results in a failure like this: [ 16.606268] fe: ASoC: fe no valid playback route I hacked the names of all components so they are unique, but even then, the problem is that snd_soc_dapm_new_dai_widgets() is called twice for the CPU DAI, once from soc_probe_platform(), and later from soc_probe_link_dais(), and both calls would create DAPM widgets with identical names, resulting in the same problem as described above. I don't know how to handle this, but I believe that nobody was really hit by this issue yet, as DPCM doesn't seem to be used widely, at least not by any machine code in mainline. One idea would be to change the automatically generated names of DAI widgets in order to make them unique, or re-factor the lookup routines. Any opinion on that? Do I miss a general consideration here? With more hacks in place, the setup somehow succeeds, but when the stream is opened, we'll hit a BUG() in sound/core/pcm_memory.c, because the FE device's DMA type is undefined. But that's another topic I'll address later. Thanks, Daniel

5 9

Re: [alsa-devel] Zoom R16
by Jason Mancine 11 Mar '14

11 Mar '14

Can't seem to compile 3.11 or 3.12 for the distro I am using (AVLInux)... tested it in Ubuntu Studio 13.10 and the quirk didn't seem to have any effect. The output of lsusb is the same, but the device doesn't show up as a sound card. On Sun, Nov 24, 2013 at 9:18 AM, Daniel Mack <daniel(a)zonque.org> wrote: > On Sun, Nov 24, 2013 at 09:03:47AM -0500, Jason Mancine wrote: > > Thanks, I will give it a try. By recent kernel do you mean latest > mainline > > or should my distro's 3.6.11 be new enough? > > Anything >= 3.11 should do. > > In further replies, please do not drop the alsa-devel mailing list. > > > Thanks, > Daniel >

3 15

[alsa-devel] Update patch for codecs alias name for Dell
by Kailang 05 Mar '14

05 Mar '14

Hi Takashi, Dell assigned more codecs for alias name. I update in attach patch. Thanks. Kailang

3 7

[alsa-devel] Unsupported Valley View 'LPE' adapter?
by Adam Williamson 27 Feb '14

27 Feb '14

Hi, folks. I recently bought a Dell Venue 8 Pro - a Bay Trail / Valley View-based Intel tablet - to try and get Fedora running on it. Among the many things I'm currently fighting :) , the sound doesn't work. snd-hda-intel is not loaded at all, and there are no audio devices present at any level. In Windows' Device Manager I see an 'Intel SST Audio Device' with PCI ID 0x86:0f28 . Indeed, the PCI ID database has this listed as "ValleyView LPE Audio Controller" - http://pci-ids.ucw.cz/read/PC/8086/0f28 . But hda-intel does not have 0f28 as a supported device. The commit for 'Bay Trail' audio support added the ID 0f04: https://lkml.org/lkml/2013/5/16/38 So...what is this 0f28? Is support for it going to be added? Is there any info I can provide to help? (I can boot Linux on the tablet and get info out of it _just barely_, but it's pretty painful - there's no wifi and an install to 'disk' (really a usb stick) doesn't work too well, so I'm stuck working in a live image with no network). -- Adam Williamson Fedora QA Community Monkey IRC: adamw | Twitter: AdamW_Fedora | XMPP: adamw AT happyassassin . net http://www.happyassassin.net

6 13

[alsa-devel] No sound from Headphone Jack over Lenovo Docking Station with T440p Laptop
by Nicolas Kalkhof 22 Feb '14

22 Feb '14

Hello, I have a Lenovo ThinkPad T440p laptop connected to an Ultra Docking Station. Sound works fine over Laptop Speakers and the 3,5mm Headphone Jack. However that is not the case when I connect the T440p to the Docking station Type 40A10065EU and use the Headphone Jack at the Dock. I've tried loading the alsa module with the "model=thinkpad" or "model=lenovo-dock" without success. My Distro is Gentoo and the Kernel is 3.13-rc4. alsa.conf: # ALSA portion alias char-major-116 snd # OSS/Free portion alias char-major-14 soundcore ## ## IMPORTANT: ## You need to customise this section for your specific sound card(s) ## and then run `update-modules' command. ## Read alsa-driver's INSTALL file in /usr/share/doc for more info. ## ## ALSA portion ## alias snd-card-0 snd-interwave ## alias snd-card-1 snd-ens1371 ## OSS/Free portion ## alias sound-slot-0 snd-card-0 ## alias sound-slot-1 snd-card-1 ## # OSS/Free portion - card #1 alias sound-service-0-0 snd-mixer-oss alias sound-service-0-1 snd-seq-oss alias sound-service-0-3 snd-pcm-oss alias sound-service-0-8 snd-seq-oss alias sound-service-0-12 snd-pcm-oss ## OSS/Free portion - card #2 ## alias sound-service-1-0 snd-mixer-oss ## alias sound-service-1-3 snd-pcm-oss ## alias sound-service-1-12 snd-pcm-oss alias /dev/mixer snd-mixer-oss alias /dev/dsp snd-pcm-oss alias /dev/midi snd-seq-oss # Set this to the correct number of cards. options snd cards_limit=1 #options snd-hda-intel model=thinkpad options snd-hda-intel model=thinkpad enable=1 index=0 options snd-hda-intel model=hdmi enable=1 index=-2 lspci -vv and dmesg goes here: [1]http://pastebin.ca/2510939 Any Help? References 1. http://pastebin.ca/2510939

4 9

[alsa-devel] [RFC v2][PATCH 00/38] Enhancement for support of some firewire devices
by Takashi Sakamoto 09 Feb '14

09 Feb '14

This series of patch is to update patches which I've posted below: [RFC][PATCH 00/17] Enhancement for firewire-lib http://mailman.alsa-project.org/pipermail/alsa-devel/2013-November/069163.h… [RFC][PATCH 00/13] a new driver for BridgeCo BeBoB based device http://mailman.alsa-project.org/pipermail/alsa-devel/2013-November/069183.h… [PATCH v2 0/8][RFC] a driver for Fireworks based devices http://mailman.alsa-project.org/pipermail/alsa-devel/2013-December/070043.h… In this time, I'll post all patches because now they have dependency each other. Each driver has lines in asound.h/firewire.h. Changes since previous RFC version of my patches: [firewire-lib] - support 32bit sample for capturing (remove 16bit sample support) - limit the number of MIDI conformant data channel by 1 - change the way to trigger MIDI messages - fix some bugs to playback/capture samples in 'dual wire' mode - fix wrong mappling of channels in 'dual wire' mode - add unit/subunit support for PLUG INFO command - add limit for dev_err() [fireworks] - improve return value for hwdep interface [bebob] - reconfirm supported devices - read root directory in config rom to get vendor name charactors - add a new way to detect clock source - remove model specific operations for PreSonus - split codes for M-Audio to three patches - remove control interface except for M-Audio special devices Best regards, Takashi Sakamoto Takashi Sakamoto (38): firewire-lib: Rename functions, structure, member for AMDTP firewire-lib: Add macros instead of fixed value for AMDTP firewire-lib: Add 'direction' member to 'amdtp_stream' structure firewire-lib: Split some codes into functions to reuse for both streams firewire-lib: Add support for AMDTP in-stream and PCM capture firewire-lib: Add support for MIDI capture/playback firewire-lib: Give syt value as parameter to handle_out_packet() firewire-lib: Add support for duplex streams synchronization in blocking mode firewire-lib: Add sort function for transmitted packet firewire-lib: Add transfer delay to synchronized duplex streams firewire-lib: Add support for channel mapping firewire-lib: Rename macros, variables and functions for CMP firewire-lib: Add 'direction' member to 'cmp_connection' structure firewire-lib: Add handling output connection by CMP firewire-lib: Add a new function to check others' connection firewire-lib: Add some AV/C general commands firewire-lib: Add quirks for Fireworks fireworks: Add skelton for Fireworks based devices fireworks: Add transaction and some commands fireworks: Add connection and stream management fireworks: Add proc interface for debugging purpose fireworks: Add MIDI interface fireworks: Add PCM interface fireworks: Add hwdep interface fireworks: Add command/response functionality into hwdep interface bebob: Add skelton for BeBoB based devices bebob: Add commands and connections/streams management bebob: Add proc interface for debugging purpose bebob: Add MIDI interface bebob: Add PCM interface bebob: Add hwdep interface bebob: Prepare for device specific operations bebob: Add support for Terratec PHASE, EWS series and Aureon bebob: Add support for Yamaha GO series bebob: Add support for Focusrite Saffire/SaffirePro series bebob: Add support for M-Audio usual Firewire series bebob: Add support for M-Audio special Firewire series bebob: Send a cue to load firmware for M-Audio Firewire series include/uapi/sound/asound.h | 4 +- include/uapi/sound/firewire.h | 20 + sound/firewire/Kconfig | 55 ++ sound/firewire/Makefile | 2 + sound/firewire/amdtp.c | 917 +++++++++++++++++------ sound/firewire/amdtp.h | 163 +++- sound/firewire/bebob/Makefile | 4 + sound/firewire/bebob/bebob.c | 430 +++++++++++ sound/firewire/bebob/bebob.h | 268 +++++++ sound/firewire/bebob/bebob_command.c | 352 +++++++++ sound/firewire/bebob/bebob_focusrite.c | 289 +++++++ sound/firewire/bebob/bebob_hwdep.c | 197 +++++ sound/firewire/bebob/bebob_maudio.c | 852 +++++++++++++++++++++ sound/firewire/bebob/bebob_midi.c | 156 ++++ sound/firewire/bebob/bebob_pcm.c | 430 +++++++++++ sound/firewire/bebob/bebob_proc.c | 185 +++++ sound/firewire/bebob/bebob_stream.c | 853 +++++++++++++++++++++ sound/firewire/bebob/bebob_terratec.c | 68 ++ sound/firewire/bebob/bebob_yamaha.c | 50 ++ sound/firewire/cmp.c | 217 ++++-- sound/firewire/cmp.h | 14 +- sound/firewire/dice.c | 46 +- sound/firewire/fcp.c | 151 ++++ sound/firewire/fcp.h | 21 + sound/firewire/fireworks/Makefile | 4 + sound/firewire/fireworks/fireworks.c | 330 ++++++++ sound/firewire/fireworks/fireworks.h | 222 ++++++ sound/firewire/fireworks/fireworks_command.c | 397 ++++++++++ sound/firewire/fireworks/fireworks_hwdep.c | 299 ++++++++ sound/firewire/fireworks/fireworks_midi.c | 166 ++++ sound/firewire/fireworks/fireworks_pcm.c | 449 +++++++++++ sound/firewire/fireworks/fireworks_proc.c | 206 +++++ sound/firewire/fireworks/fireworks_stream.c | 354 +++++++++ sound/firewire/fireworks/fireworks_transaction.c | 327 ++++++++ sound/firewire/speakers.c | 94 +-- 35 files changed, 8196 insertions(+), 396 deletions(-) create mode 100644 sound/firewire/bebob/Makefile create mode 100644 sound/firewire/bebob/bebob.c create mode 100644 sound/firewire/bebob/bebob.h create mode 100644 sound/firewire/bebob/bebob_command.c create mode 100644 sound/firewire/bebob/bebob_focusrite.c create mode 100644 sound/firewire/bebob/bebob_hwdep.c create mode 100644 sound/firewire/bebob/bebob_maudio.c create mode 100644 sound/firewire/bebob/bebob_midi.c create mode 100644 sound/firewire/bebob/bebob_pcm.c create mode 100644 sound/firewire/bebob/bebob_proc.c create mode 100644 sound/firewire/bebob/bebob_stream.c create mode 100644 sound/firewire/bebob/bebob_terratec.c create mode 100644 sound/firewire/bebob/bebob_yamaha.c create mode 100644 sound/firewire/fireworks/Makefile create mode 100644 sound/firewire/fireworks/fireworks.c create mode 100644 sound/firewire/fireworks/fireworks.h create mode 100644 sound/firewire/fireworks/fireworks_command.c create mode 100644 sound/firewire/fireworks/fireworks_hwdep.c create mode 100644 sound/firewire/fireworks/fireworks_midi.c create mode 100644 sound/firewire/fireworks/fireworks_pcm.c create mode 100644 sound/firewire/fireworks/fireworks_proc.c create mode 100644 sound/firewire/fireworks/fireworks_stream.c create mode 100644 sound/firewire/fireworks/fireworks_transaction.c -- 1.8.3.2

4 50

[alsa-devel] [PATCH v3] ASoC: add RT286 CODEC driver
by bardliao＠realtek.com 07 Feb '14

07 Feb '14

From: Bard Liao <bardliao(a)realtek.com> This patch adds the ALC286 codec driver. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- * Initialize all cache registers in rt286_reg_cache, and remove rt286_reg_cache size define. --- include/sound/rt286.h | 19 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt286.c | 1373 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/rt286.h | 89 +++ 5 files changed, 1487 insertions(+) create mode 100644 include/sound/rt286.h create mode 100644 sound/soc/codecs/rt286.c create mode 100644 sound/soc/codecs/rt286.h diff --git a/include/sound/rt286.h b/include/sound/rt286.h new file mode 100644 index 0000000..6c0c1c9 --- /dev/null +++ b/include/sound/rt286.h @@ -0,0 +1,19 @@ +/* + * linux/sound/rt286.h -- Platform data for RT286 + * + * Copyright 2013 Realtek Microelectronics + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __LINUX_SND_RT286_H +#define __LINUX_SND_RT286_H + +struct rt286_platform_data { + bool cbj_en; /*combo jack enable*/ + bool irq_en; /*IRQ enable*/ +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 983d087a..755a39d 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -61,6 +61,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_PCM3008 select SND_SOC_RT5631 if I2C select SND_SOC_RT5640 if I2C + select SND_SOC_RT286 if I2C select SND_SOC_SGTL5000 if I2C select SND_SOC_SI476X if MFD_SI476X_CORE select SND_SOC_SN95031 if INTEL_SCU_IPC @@ -319,6 +320,9 @@ config SND_SOC_RT5631 config SND_SOC_RT5640 tristate +config SND_SOC_RT286 + tristate + #Freescale sgtl5000 codec config SND_SOC_SGTL5000 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index bc12676..b31615c 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -48,6 +48,7 @@ snd-soc-pcm1792a-codec-objs := pcm1792a.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-rt5631-objs := rt5631.o snd-soc-rt5640-objs := rt5640.o +snd-soc-rt286-objs := rt286.o snd-soc-sgtl5000-objs := sgtl5000.o snd-soc-alc5623-objs := alc5623.o snd-soc-alc5632-objs := alc5632.o @@ -181,6 +182,7 @@ obj-$(CONFIG_SND_SOC_PCM1792A) += snd-soc-pcm1792a-codec.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o +obj-$(CONFIG_SND_SOC_RT286) += snd-soc-rt286.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o obj-$(CONFIG_SND_SOC_SIGMADSP) += snd-soc-sigmadsp.o obj-$(CONFIG_SND_SOC_SI476X) += snd-soc-si476x.o diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c new file mode 100644 index 0000000..214e5fc --- /dev/null +++ b/sound/soc/codecs/rt286.c @@ -0,0 +1,1373 @@ +/* + * rt286.c -- RT286 ALSA SoC audio codec driver + * + * Copyright 2013 Realtek Semiconductor Corp. + * Author: Bard Liao <bardliao(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/spi/spi.h> +#include <linux/acpi.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <sound/rt286.h> + +#include "../../pci/hda/hda_codec.h" +#include "rt286.h" + +struct rt286_priv { + struct regmap *regmap; + struct rt286_platform_data pdata; + int sys_clk; +}; + +static unsigned int rt286_reg_cache[] = { + [NODE_ID_AUDIO_FUNCTION_GROUP] = 0x0000, + [NODE_ID_DAC_OUT1] = 0x7f7f, + [NODE_ID_DAC_OUT2] = 0x7f7f, + [NODE_ID_SPDIF] = 0x0000, + [NODE_ID_ADC_IN1] = 0x4343, + [NODE_ID_ADC_IN2] = 0x4343, + [NODE_ID_MIC1] = 0x0000, + [NODE_ID_MIXER_IN] = 0x000b, + [NODE_ID_MIXER_OUT1] = 0x0002, + [NODE_ID_MIXER_OUT2] = 0x0000, + [NODE_ID_DMIC1] = 0x0000, + [NODE_ID_DMIC2] = 0x0000, + [NODE_ID_LINE1] = 0x0000, + [NODE_ID_BEEP] = 0x0000, + [NODE_ID_VENDOR_REGISTERS] = 0x0000, + [NODE_ID_SPK_OUT] = 0x0000, + [NODE_ID_HP_OUT] = 0x0000, + [NODE_ID_MIXER_IN1] = 0x0000, + [NODE_ID_MIXER_IN2] = 0x0000, +}; + +static int rt286_hw_read(void *context, unsigned int reg, unsigned int *value) +{ + struct i2c_client *client = context; + struct i2c_msg xfer[2]; + int ret; + unsigned int buf = 0x0; + + if (reg <= 0xff) { /*read cache*/ + *value = rt286_reg_cache[reg]; + return 0; + } + reg = cpu_to_be32(reg); + /* Write register */ + xfer[0].addr = client->addr; + xfer[0].flags = 0; + xfer[0].len = 4; + xfer[0].buf = (u8 *)® + + /* Read data */ + xfer[1].addr = client->addr; + xfer[1].flags = I2C_M_RD; + xfer[1].len = 4; + xfer[1].buf = (u8 *)&buf; + + ret = i2c_transfer(client->adapter, xfer, 2); + if (ret < 0) + return ret; + else if (ret != 2) + return -EIO; + + *value = be32_to_cpu(buf); + + return 0; +} + +static int rt286_hw_write(void *context, unsigned int reg, unsigned int value) +{ + struct i2c_client *client = context; + u8 data[4]; + int ret; + + if (reg <= 0xff) { /*write cache*/ + rt286_reg_cache[reg] = value; + return 0; + } + data[0] = (reg >> 24) & 0xff; + data[1] = (reg >> 16) & 0xff; + data[2] = (reg >> 8) & 0xff; + data[3] = reg & 0xff; + + ret = i2c_master_send(client, data, 4); + if (ret == 4) + return 0; + if (ret < 0) + return ret; + else + return -EIO; +} + +static unsigned int rt286_read(struct snd_soc_codec *codec, + unsigned int vid, + unsigned int nid, + unsigned int data) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + unsigned int verb; + unsigned int buf = 0x0; + + verb = (nid << 20) | (vid << 8) | data; + + regmap_read(rt286->regmap, verb, &buf); + return buf; +} + +static int rt286_write(struct snd_soc_codec *codec, + unsigned int vid, + unsigned int nid, + unsigned int value) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + unsigned int reg; + + reg = (nid << 20) | (vid << 8) | value; + + return regmap_write(rt286->regmap, reg, 0); +} + +static int rt286_update_bits(struct snd_soc_codec *codec, unsigned int vid, + unsigned int nid, unsigned int data, + unsigned int mask, unsigned int value) +{ + unsigned int old, new; + int change, ret; + + old = rt286_read(codec, vid | 0x800, nid, data); + new = (old & ~mask) | (value & mask); + change = old != new; + + if (change) { + ret = rt286_write(codec, vid, nid, new); + if (ret < 0) { + dev_err(codec->dev, + "Failed to write private reg: %d\n", ret); + goto err; + } + } + return change; + +err: + return ret; +} + +static int rt286_index_write(struct snd_soc_codec *codec, + unsigned int wid, unsigned int index, unsigned int data) +{ + int ret; + + ret = rt286_write(codec, AC_VERB_SET_COEF_INDEX, + wid, index); + if (ret < 0) { + dev_err(codec->dev, "Failed to set private addr: %d\n", ret); + goto err; + } + ret = rt286_write(codec, AC_VERB_SET_PROC_COEF, + wid, data); + if (ret < 0) { + dev_err(codec->dev, "Failed to set private value: %d\n", ret); + goto err; + } + + return 0; +err: + return ret; +} + +static unsigned int rt286_index_read(struct snd_soc_codec *codec, + unsigned int wid, unsigned int index) +{ + int ret; + + ret = rt286_write(codec, AC_VERB_SET_COEF_INDEX, + wid, index); + if (ret < 0) { + dev_err(codec->dev, "Failed to set private addr: %d\n", ret); + return ret; + } + return rt286_read(codec, AC_VERB_GET_PROC_COEF, + wid, index); +} + +static int rt286_index_update_bits(struct snd_soc_codec *codec, + unsigned int wid, unsigned int index, + unsigned int mask, unsigned int data) +{ + unsigned int old, new; + int change, ret; + + old = rt286_index_read(codec, wid, index); + new = (old & ~mask) | (data & mask); + change = old != new; + + if (change) { + ret = rt286_index_write(codec, wid, index, new); + if (ret < 0) { + dev_err(codec->dev, + "Failed to write private reg: %d\n", ret); + goto err; + } + } + return change; + +err: + return ret; +} + +static int rt286_support_power_controls[] = { + NODE_ID_DAC_OUT1, + NODE_ID_DAC_OUT2, + NODE_ID_ADC_IN1, + NODE_ID_ADC_IN2, + NODE_ID_MIC1, + NODE_ID_DMIC1, + NODE_ID_DMIC2, + NODE_ID_SPK_OUT, + NODE_ID_HP_OUT, +}; +#define RT286_POWER_REG_LEN ARRAY_SIZE(rt286_support_power_controls) + +static const struct reg_default rt286_reg[] = { + { NODE_ID_DAC_OUT1, 0x7f7f }, + { NODE_ID_DAC_OUT2, 0x7f7f }, + { NODE_ID_SPDIF, 0x0000 }, + { NODE_ID_ADC_IN1, 0x4343 }, + { NODE_ID_ADC_IN2, 0x4343 }, + { NODE_ID_MIC1, 0x0000 }, + { NODE_ID_MIXER_IN, 0x000b }, + { NODE_ID_MIXER_OUT1, 0x0002 }, + { NODE_ID_MIXER_OUT2, 0x0000 }, + { NODE_ID_SPK_OUT, 0x0000 }, + { NODE_ID_HP_OUT, 0x0000 }, + { NODE_ID_MIXER_IN1, 0x0005 }, + { NODE_ID_MIXER_IN2, 0x0005 }, +}; + +static bool rt286_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NODE_ID_DAC_OUT1: + case NODE_ID_DAC_OUT2: + case NODE_ID_ADC_IN1: + case NODE_ID_ADC_IN2: + case NODE_ID_MIC1: + case NODE_ID_SPDIF: + case NODE_ID_MIXER_IN: + case NODE_ID_MIXER_OUT1: + case NODE_ID_MIXER_OUT2: + case NODE_ID_DMIC1: + case NODE_ID_DMIC2: + case NODE_ID_SPK_OUT: + case NODE_ID_HP_OUT: + case NODE_ID_MIXER_IN1: + case NODE_ID_MIXER_IN2: + return true; + default: + return false; + } +} + +int rt286_jack_detect(struct snd_soc_codec *codec, bool *hp, bool *mic) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + unsigned int val; + int i; + + *hp = false; + *mic = false; + + if (rt286->pdata.cbj_en) { + *hp = rt286_read(codec, AC_VERB_GET_PIN_SENSE, + NODE_ID_HP_OUT, 0); + if (*hp) { + /* power on HV,VERF */ + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x03, 0x1001, 0x0); + /* power LDO1 */ + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x08, 0x4, 0x4); + rt286_write(codec, AC_VERB_SET_PIN_WIDGET_CONTROL, + NODE_ID_MIC1, 0x24); + rt286_write(codec, AC_VERB_SET_COEF_INDEX, + 0x20, 0x50); + val = rt286_read(codec, AC_VERB_GET_PROC_COEF, + 0x20, 0x50); + + msleep(200); + i = 40; + while (((val & 0x0800) == 0) && (i > 0)) { + rt286_write(codec, AC_VERB_SET_COEF_INDEX, + 0x20, 0x50); + val = rt286_read(codec, + AC_VERB_GET_PROC_COEF, 0x20, 0x50); + i--; + msleep(20); + } + + if (0x0400 == (val & 0x0700)) { + *mic = false; + + rt286_write(codec, + AC_VERB_SET_PIN_WIDGET_CONTROL, + NODE_ID_MIC1, 0x20); + /* power off HV,VERF */ + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, + 0x03, 0x1001, 0x1001); + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, + 0x04, 0xC000, 0x0000); + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, + 0x4f, 0x0030, 0x0000); + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, + 0x02, 0xc000, 0x0000); + } else if ((0x0200 == (val & 0x0700)) || + (0x0100 == (val & 0x0700))) { + *mic = true; + } else { + *mic = false; + } + + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, + 0x20, 0x0060, 0x0000); + } else { + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, + 0x20, 0x0060, 0x0020); + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, + 0x04, 0xC000, 0x8000); + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, + 0x4f, 0x0030, 0x0020); + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, + 0x02, 0xc000, 0x8000); + + *mic = false; + } + } else { + *hp = rt286_read(codec, AC_VERB_GET_PIN_SENSE, + NODE_ID_HP_OUT, 0); + *mic = rt286_read(codec, AC_VERB_GET_PIN_SENSE, + NODE_ID_MIC1, 0); + } + + + /* Clear IRQ */ + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x33, 0x1, 0x1); + return 0; +} +EXPORT_SYMBOL_GPL(rt286_jack_detect); + +static const DECLARE_TLV_DB_SCALE(out_vol_tlv, -6350, 50, 0); +static const DECLARE_TLV_DB_SCALE(mic_vol_tlv, 0, 1000, 0); + +static int rt286_vol_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int io = 0x8000; + + if (NODE_ID_ADC_IN1 == mc->reg) + io = 0; + ucontrol->value.integer.value[0] = + rt286_read(codec, AC_VERB_GET_AMP_GAIN_MUTE, + mc->reg, 0x2000 | io) & 0x7f; + ucontrol->value.integer.value[1] = + rt286_read(codec, AC_VERB_GET_AMP_GAIN_MUTE, + mc->reg, io) & 0x7f; + + return 0; +} + +static int rt286_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int vall, valr, io; + + if (NODE_ID_ADC_IN1 == mc->reg) + io = 0x4000; + else + io = 0x8000; + vall = ucontrol->value.integer.value[0]; + valr = ucontrol->value.integer.value[1]; + if (vall == valr) { + vall = vall | 0x3000 | io; + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, mc->reg, vall); + } else { + vall = vall | 0x2000 | io; + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, mc->reg, vall); + valr = valr | 0x1000 | io; + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, mc->reg, valr); + } + + return 0; +} + +static int rt286_mic_gain_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + ucontrol->value.integer.value[0] = + rt286_read(codec, AC_VERB_GET_AMP_GAIN_MUTE, + mc->reg, 0x0) & 0x3; + + return 0; +} + +static int rt286_mic_gain_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int val; + + val = ucontrol->value.integer.value[0]; + val = val | 0x7000; + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, mc->reg, val); + + return 0; +} + +static int rt286_playback_switch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + + + ucontrol->value.integer.value[0] = + !(rt286_read(codec, AC_VERB_GET_AMP_GAIN_MUTE, + mc->reg, 0xa000) & 0x80); + ucontrol->value.integer.value[1] = + !(rt286_read(codec, AC_VERB_GET_AMP_GAIN_MUTE, + mc->reg, 0x8000) & 0x80); + + return 0; +} + +static int rt286_playback_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct soc_mixer_control *mc = + (struct soc_mixer_control *)kcontrol->private_value; + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + unsigned int vall, valr, val = 0; + + vall = (!ucontrol->value.integer.value[0] << 7); + valr = (!ucontrol->value.integer.value[1] << 7); + if (vall) + val |= 0x8000; + if (valr) + val |= 0x0080; + + if (vall == valr) { + vall = vall | 0xb000; + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, mc->reg, vall); + } else { + vall = vall | 0xa000; + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, mc->reg, vall); + valr = valr | 0x9000; + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, mc->reg, valr); + } + snd_soc_update_bits(codec, mc->reg, 0x8080, val); + + return 0; +} + +static const struct snd_kcontrol_new rt286_snd_controls[] = { + SOC_DOUBLE_EXT_TLV("DAC0 Playback Volume", NODE_ID_DAC_OUT1, + 8, 0, 0x7f, 0, + rt286_vol_get, rt286_vol_put, + out_vol_tlv), + SOC_DOUBLE_EXT_TLV("ADC0 Capture Volume", NODE_ID_ADC_IN1, + 8, 0, 0x7f, 0, + rt286_vol_get, rt286_vol_put, + out_vol_tlv), + SOC_SINGLE_EXT_TLV("AMIC Volume", NODE_ID_MIC1, 0, 0x3, 0, + rt286_mic_gain_get, rt286_mic_gain_put, + mic_vol_tlv), + SOC_DOUBLE_EXT("Headphone Playback Switch", NODE_ID_HP_OUT, + 15, 8, 1, 1, rt286_playback_switch_get, + rt286_playback_switch_put), + SOC_DOUBLE_EXT("Speaker Playback Switch", NODE_ID_SPK_OUT, + 15, 8, 1, 1, rt286_playback_switch_get, + rt286_playback_switch_put), +}; + +static void hw_configure(struct snd_soc_codec *codec) +{ + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + int i; + + rt286_write(codec, AC_VERB_SET_POWER_STATE, + NODE_ID_AUDIO_FUNCTION_GROUP, AC_PWRST_D3); + for (i = 0; i < RT286_POWER_REG_LEN; i++) { + rt286_write(codec, AC_VERB_SET_POWER_STATE, + rt286_support_power_controls[i], AC_PWRST_D3); + } + + if (rt286->pdata.irq_en) + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x33, 0x0002, 0x0002); + + if (!rt286->pdata.cbj_en) { + rt286_index_write(codec, + NODE_ID_VENDOR_REGISTERS, 0x50, 0x0000); + rt286_index_write(codec, + NODE_ID_VENDOR_REGISTERS, 0x19, 0x0816); + rt286_index_write(codec, + NODE_ID_VENDOR_REGISTERS, 0x20, 0x0000); + } + + mdelay(10); + + /* Set configuration default */ + rt286_write(codec, AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, + NODE_ID_MIC1, 0x00); + rt286_write(codec, AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, + NODE_ID_DMIC1, 0x00); + rt286_write(codec, AC_VERB_SET_CONFIG_DEFAULT_BYTES_3, + NODE_ID_DMIC2, 0x00); + + mdelay(10); + + if (rt286->pdata.cbj_en) + rt286_index_write(codec, + NODE_ID_VENDOR_REGISTERS, 0x4f, 0x5029); + else + rt286_index_write(codec, + NODE_ID_VENDOR_REGISTERS, 0x4f, 0xb029); + + /*Power down LDO2*/ + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x08, 0x0008, 0x0000); +} + +/* Digital Mixer */ +static const struct snd_kcontrol_new rt286_front_mix[] = { + SOC_DAPM_SINGLE("DAC Switch", NODE_ID_MIXER_OUT1, + RT286_M_FRONT_DAC_SFT, 1, 1), + SOC_DAPM_SINGLE("RECMIX Switch", NODE_ID_MIXER_OUT1, + RT286_M_FRONT_REC_SFT, 1, 1), +}; + +/* Analog Input Mixer */ +static const struct snd_kcontrol_new rt286_rec_mix[] = { + SOC_DAPM_SINGLE("Beep Switch", NODE_ID_MIXER_IN, + RT286_M_REC_BEEP_SFT, 1, 1), + SOC_DAPM_SINGLE("Line1 Switch", NODE_ID_MIXER_IN, + RT286_M_REC_LINE1_SFT, 1, 1), + SOC_DAPM_SINGLE("Mic1 Switch", NODE_ID_MIXER_IN, + RT286_M_REC_MIC1_SFT, 1, 1), + SOC_DAPM_SINGLE("I2S Switch", NODE_ID_MIXER_IN, + RT286_M_REC_I2S_SFT, 1, 1), +}; + +/* ADC0 source */ +static const char * const rt286_adc_src[] = { + "Mic", "Dmic" +}; + +static int rt286_adc_values[] = { + 0, 5, +}; + +static const SOC_VALUE_ENUM_SINGLE_DECL( + rt286_adc0_enum, NODE_ID_MIXER_IN1, RT286_ADC_SEL_SFT, + RT286_ADC_SEL_MASK, rt286_adc_src, rt286_adc_values); + +static const struct snd_kcontrol_new rt286_adc0_mux = + SOC_DAPM_VALUE_ENUM("ADC 0 source", rt286_adc0_enum); + +static const SOC_VALUE_ENUM_SINGLE_DECL( + rt286_adc1_enum, NODE_ID_MIXER_IN2, RT286_ADC_SEL_SFT, + RT286_ADC_SEL_MASK, rt286_adc_src, rt286_adc_values); + +static const struct snd_kcontrol_new rt286_adc1_mux = + SOC_DAPM_VALUE_ENUM("ADC 1 source", rt286_adc1_enum); + +/* HP-OUT source */ +static const char * const rt286_hpo_src[] = { + "Front", "Surround" +}; + +static const SOC_ENUM_SINGLE_DECL(rt286_hpo_enum, NODE_ID_HP_OUT, + RT286_HP_SEL_SFT, rt286_hpo_src); + +static const struct snd_kcontrol_new rt286_hpo_mux = +SOC_DAPM_ENUM("HPO source", rt286_hpo_enum); + +/* SPK-OUT source */ +static const char * const rt286_spo_src[] = { + "Front", "Surround" +}; + +static const SOC_ENUM_SINGLE_DECL(rt286_spo_enum, NODE_ID_SPK_OUT, + RT286_SPK_SEL_SFT, rt286_spo_src); + +static const struct snd_kcontrol_new rt286_spo_mux = +SOC_DAPM_ENUM("SPO source", rt286_spo_enum); + +/* SPDIF source */ +static const char * const rt286_spdif_src[] = { + "PCM-IN 0", "PCM-IN 1", "SP-OUT", "PP" +}; + +static const SOC_ENUM_SINGLE_DECL(rt286_spdif_enum, NODE_ID_SPDIF, + RT286_SPDIF_SEL_SFT, rt286_spdif_src); + +static const struct snd_kcontrol_new rt286_spdif_mux = +SOC_DAPM_ENUM("SPDIF source", rt286_spdif_enum); + + +static int rt286_spk_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + rt286_write(codec, AC_VERB_SET_POWER_STATE, + NODE_ID_SPK_OUT, AC_PWRST_D0); + rt286_write(codec, AC_VERB_SET_PIN_WIDGET_CONTROL, + NODE_ID_SPK_OUT, 0x40); + rt286_write(codec, AC_VERB_SET_EAPD_BTLENABLE, + NODE_ID_SPK_OUT, 0x0002); + break; + case SND_SOC_DAPM_PRE_PMD: + rt286_write(codec, AC_VERB_SET_EAPD_BTLENABLE, + NODE_ID_SPK_OUT, 0x0000); + rt286_write(codec, AC_VERB_SET_PIN_WIDGET_CONTROL, + NODE_ID_SPK_OUT, 0x00); + rt286_write(codec, AC_VERB_SET_POWER_STATE, + NODE_ID_SPK_OUT, AC_PWRST_D3); + break; + + default: + return 0; + } + + return 0; +} + +static int rt286_hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + unsigned int val; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + val = snd_soc_read(codec, NODE_ID_HP_OUT) & 0x8080; + switch (val) { + case 0x0: + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, + NODE_ID_HP_OUT, 0xb000); + break; + case 0x8000: + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, + NODE_ID_HP_OUT, 0x9000); + break; + case 0x0080: + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, + NODE_ID_HP_OUT, 0xa000); + break; + } + if (val != 0x8080) + mdelay(100); + break; + + case SND_SOC_DAPM_PRE_PMD: + rt286_write(codec, AC_VERB_SET_AMP_GAIN_MUTE, + NODE_ID_HP_OUT, 0xb080); + break; + + default: + return 0; + } + + return 0; +} + +static int rt286_set_power_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + rt286_write(codec, AC_VERB_SET_POWER_STATE, + w->reg, AC_PWRST_D0); + break; + case SND_SOC_DAPM_PRE_PMD: + rt286_write(codec, AC_VERB_SET_POWER_STATE, + w->reg, AC_PWRST_D3); + break; + default: + return 0; + } + + return 0; +} + +static int rt286_set_dmic1_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + rt286_write(codec, AC_VERB_SET_PIN_WIDGET_CONTROL, + NODE_ID_DMIC1, 0x20); + rt286_write(codec, AC_VERB_SET_POWER_STATE, + NODE_ID_DMIC1, AC_PWRST_D0); + break; + case SND_SOC_DAPM_PRE_PMD: + rt286_write(codec, AC_VERB_SET_POWER_STATE, + NODE_ID_DMIC1, AC_PWRST_D3); + rt286_write(codec, AC_VERB_SET_PIN_WIDGET_CONTROL, + NODE_ID_DMIC1, 0x0); + break; + default: + return 0; + } + + return 0; +} + +static int rt286_hp_pow_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + rt286_write(codec, AC_VERB_SET_POWER_STATE, + NODE_ID_HP_OUT, AC_PWRST_D0); + rt286_write(codec, AC_VERB_SET_PIN_WIDGET_CONTROL, + NODE_ID_HP_OUT, 0x40); + break; + + case SND_SOC_DAPM_PRE_PMD: + rt286_write(codec, AC_VERB_SET_PIN_WIDGET_CONTROL, + NODE_ID_HP_OUT, 0x00); + rt286_write(codec, AC_VERB_SET_POWER_STATE, + NODE_ID_HP_OUT, AC_PWRST_D3); + break; + + default: + return 0; + } + + return 0; +} + +static int rt286_adc_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + struct soc_enum *e; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (rt286->pdata.cbj_en) { + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x04, 0xC000, 0x8000); + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x4f, 0x0030, 0x0020); + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x02, 0xc000, 0x8000); + } + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x0f, 0x0020, 0x0000); + rt286_write(codec, AC_VERB_SET_POWER_STATE, + w->reg, AC_PWRST_D0); + rt286_update_bits(codec, AC_VERB_SET_AMP_GAIN_MUTE, + w->reg, 0, 0x7080, 0x7000); + break; + case SND_SOC_DAPM_PRE_PMD: + rt286_update_bits(codec, AC_VERB_SET_AMP_GAIN_MUTE, + w->reg, 0, 0x7080, 0x7080); + rt286_write(codec, AC_VERB_SET_POWER_STATE, + w->reg, AC_PWRST_D3); + break; + case SND_SOC_DAPM_POST_REG: + e = (struct soc_enum *)kcontrol->private_value; + rt286_write(codec, AC_VERB_SET_CONNECT_SEL, e->reg, + snd_soc_read(codec, e->reg) & RT286_ADC_SEL_MASK); + break; + default: + return 0; + } + + return 0; +} + +static int rt286_out_mux_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + + switch (event) { + case SND_SOC_DAPM_POST_REG: + rt286_write(codec, AC_VERB_SET_CONNECT_SEL, + e->reg, snd_soc_read(codec, e->reg) & 0x1); + break; + default: + return 0; + } + + return 0; +} + +static const struct snd_soc_dapm_widget rt286_dapm_widgets[] = { + /* Input Lines */ + SND_SOC_DAPM_INPUT("DMIC1 Pin"), + SND_SOC_DAPM_INPUT("DMIC2 Pin"), + SND_SOC_DAPM_INPUT("MIC1"), + SND_SOC_DAPM_INPUT("LINE1"), + SND_SOC_DAPM_INPUT("Beep"), + + /* DMIC */ + SND_SOC_DAPM_PGA_E("DMIC1", NODE_ID_DMIC1, 15, 0, + NULL, 0, rt286_set_dmic1_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_E("DMIC2", NODE_ID_DMIC2, 15, 0, + NULL, 0, rt286_set_power_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_SUPPLY("DMIC Receiver", SND_SOC_NOPM, + 0, 0, NULL, 0), + + /* REC Mixer */ + SND_SOC_DAPM_MIXER("RECMIX", SND_SOC_NOPM, 0, 0, + rt286_rec_mix, ARRAY_SIZE(rt286_rec_mix)), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC 0", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC 1", NULL, SND_SOC_NOPM, 0, 0), + + /* ADC Mux */ + SND_SOC_DAPM_MUX_E("ADC 0 Mux", NODE_ID_ADC_IN1, 15, 0, + &rt286_adc0_mux, rt286_adc_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_REG), + SND_SOC_DAPM_MUX_E("ADC 1 Mux", NODE_ID_ADC_IN2, 15, 0, + &rt286_adc1_mux, rt286_adc_event, SND_SOC_DAPM_PRE_PMD | + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_REG), + + + /* Audio Interface */ + SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_IN("AIF2RX", "AIF2 Playback", 0, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_AIF_OUT("AIF2TX", "AIF2 Capture", 0, SND_SOC_NOPM, 0, 0), + + + /* Output Side */ + + /* DACs */ + SND_SOC_DAPM_DAC_E("DAC 0", NULL, NODE_ID_DAC_OUT1, + 15, 0, rt286_set_power_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_DAC_E("DAC 1", NULL, NODE_ID_DAC_OUT2, + 15, 0, rt286_set_power_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + + /* Output Mux */ + SND_SOC_DAPM_MUX_E("SPK Mux", SND_SOC_NOPM, 0, 0, + &rt286_spo_mux, + rt286_out_mux_event, + SND_SOC_DAPM_POST_REG), + + SND_SOC_DAPM_MUX_E("HPO Mux", SND_SOC_NOPM, 0, 0, + &rt286_hpo_mux, + rt286_out_mux_event, + SND_SOC_DAPM_POST_REG), + + SND_SOC_DAPM_SUPPLY("HP Power", SND_SOC_NOPM, + 0, 0, rt286_hp_pow_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + + SND_SOC_DAPM_MUX("SPDIF Mux", SND_SOC_NOPM, 0, 0, + &rt286_spdif_mux), + + /* Output Mixer */ + SND_SOC_DAPM_MIXER("Front", SND_SOC_NOPM, 0, 0, + rt286_front_mix, ARRAY_SIZE(rt286_front_mix)), + SND_SOC_DAPM_PGA("Surround", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Output Pga */ + SND_SOC_DAPM_PGA_E("SPO", SND_SOC_NOPM, 0, 0, + NULL, 0, rt286_spk_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PGA_E("HPO", SND_SOC_NOPM, 0, 0, + NULL, 0, rt286_hp_event, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("SPOL"), + SND_SOC_DAPM_OUTPUT("SPOR"), + SND_SOC_DAPM_OUTPUT("HPO Pin"), + SND_SOC_DAPM_OUTPUT("SPDIF"), +}; + +static const struct snd_soc_dapm_route rt286_dapm_routes[] = { + {"DMIC1", NULL, "DMIC1 Pin"}, + {"DMIC2", NULL, "DMIC2 Pin"}, + {"DMIC1", NULL, "DMIC Receiver"}, + {"DMIC2", NULL, "DMIC Receiver"}, + + {"RECMIX", "Beep Switch", "Beep"}, + {"RECMIX", "Line1 Switch", "LINE1"}, + {"RECMIX", "Mic1 Switch", "MIC1"}, + + {"ADC 0 Mux", "Dmic", "DMIC1"}, + {"ADC 0 Mux", "Mic", "MIC1"}, + {"ADC 1 Mux", "Dmic", "DMIC2"}, + {"ADC 1 Mux", "Mic", "MIC1"}, + + {"ADC 0", NULL, "ADC 0 Mux"}, + {"ADC 1", NULL, "ADC 1 Mux"}, + + {"AIF1TX", NULL, "ADC 0"}, + {"AIF2TX", NULL, "ADC 1"}, + + {"DAC 0", NULL, "AIF1RX"}, + {"DAC 1", NULL, "AIF2RX"}, + + {"Front", "DAC Switch", "DAC 0"}, + {"Front", "RECMIX Switch", "RECMIX"}, + + {"Surround", NULL, "DAC 1"}, + + {"SPK Mux", "Front", "Front"}, + {"SPK Mux", "Surround", "Surround"}, + + {"HPO Mux", "Front", "Front"}, + {"HPO Mux", "Surround", "Surround"}, + + {"SPO", NULL, "SPK Mux"}, + {"HPO", NULL, "HPO Mux"}, + {"HPO", NULL, "HP Power"}, + + {"SPOL", NULL, "SPO"}, + {"SPOR", NULL, "SPO"}, + {"HPO Pin", NULL, "HPO"}, +}; + +static int rt286_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + unsigned int val = 0; + int d_len_code; + + switch (params_rate(params)) { + /* bit 14 0:48K 1:44.1K */ + case 44100: + val |= 0x4000; + break; + case 48000: + break; + default: + dev_err(codec->dev, "Unsupported sample rate %d\n", + params_rate(params)); + return -EINVAL; + } + switch (rt286->sys_clk) { + case 12288000: + case 24576000: + if (params_rate(params) != 48000) { + dev_err(codec->dev, "Sys_clk is not matched (%d %d)\n", + params_rate(params), rt286->sys_clk); + return -EINVAL; + } + break; + case 11289600: + case 22579200: + if (params_rate(params) != 44100) { + dev_err(codec->dev, "Sys_clk is not matched (%d %d)\n", + params_rate(params), rt286->sys_clk); + return -EINVAL; + } + break; + } + + if (params_channels(params) <= 16) { + /* bit 3:0 Number of Channel */ + val |= (params_channels(params) - 1); + } else { + dev_err(codec->dev, "Unsupported channels %d\n", + params_channels(params)); + return -EINVAL; + } + + d_len_code = 0; + switch (params_format(params)) { + /* bit 6:4 Bits per Sample */ + case SNDRV_PCM_FORMAT_S16_LE: + d_len_code = 0; + val |= (0x1 << 4); + break; + case SNDRV_PCM_FORMAT_S32_LE: + d_len_code = 2; + val |= (0x4 << 4); + break; + case SNDRV_PCM_FORMAT_S20_3LE: + d_len_code = 1; + val |= (0x2 << 4); + break; + case SNDRV_PCM_FORMAT_S24_LE: + d_len_code = 2; + val |= (0x3 << 4); + break; + case SNDRV_PCM_FORMAT_S8: + break; + default: + return -EINVAL; + } + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x09, 0x0018, d_len_code << 3); + pr_debug("format val = 0x%lx\n", val); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + rt286_update_bits(codec, AC_VERB_SET_STREAM_FORMAT, + NODE_ID_DAC_OUT1, 0, 0x407f, val); + else + rt286_update_bits(codec, AC_VERB_SET_STREAM_FORMAT, + NODE_ID_ADC_IN1, 0, 0x407f, val); + + return 0; +} + +static int rt286_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x09, 0x0800, 0x800); + break; + case SND_SOC_DAIFMT_CBS_CFS: + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x09, 0x0800, 0x0); + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x09, 0x0300, 0x0); + break; + case SND_SOC_DAIFMT_LEFT_J: + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x09, 0x0300, 0x1 << 8); + break; + case SND_SOC_DAIFMT_DSP_A: + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x09, 0x0300, 0x2 << 8); + break; + case SND_SOC_DAIFMT_DSP_B: + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x09, 0x0300, 0x3 << 8); + break; + default: + return -EINVAL; + } + /* bit 15 Stream Type 0:PCM 1:Non-PCM */ + rt286_update_bits(codec, AC_VERB_SET_STREAM_FORMAT, + NODE_ID_DAC_OUT1, 0, 0x8000, 0x0); + rt286_update_bits(codec, AC_VERB_SET_STREAM_FORMAT, + NODE_ID_ADC_IN1, 0, 0x8000, 0x0); + + return 0; +} + +static int rt286_set_dai_sysclk(struct snd_soc_dai *dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct rt286_priv *rt286 = snd_soc_codec_get_drvdata(codec); + + pr_debug("%s freq=%d\n", __func__, freq); + + if (RT286_SCLK_S_MCLK == clk_id) { + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x0a, 0x0100, 0x0); + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x49, 0x0020, 0x0020); + } else { + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x0a, 0x0100, 0x0100); + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x63, 0x4, 0x4); + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x49, 0x20, 0x0); + } + + switch (freq) { + case 19200000: + case 24000000: + if (RT286_SCLK_S_MCLK == clk_id) { + dev_err(codec->dev, "Should not use MCLK\n"); + return -EINVAL; + } + break; + case 12288000: + case 11289600: + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x0a, 0x8, 0x0); + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x0b, 0xfc1e, 0x0004); + break; + case 24576000: + case 22579200: + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x0a, 0x8, 0x8); + rt286_index_update_bits(codec, NODE_ID_VENDOR_REGISTERS, + 0x0b, 0xfc1e, 0x5406); + break; + default: + dev_err(codec->dev, "Unsupported system clock\n"); + return -EINVAL; + } + + rt286->sys_clk = freq; + + return 0; +} + +static int rt286_set_dai_dfs(struct snd_soc_dai *dai, unsigned int fs) +{ + struct snd_soc_codec *codec = dai->codec; + + pr_debug("%s fs=%d\n", __func__, fs); + if (50 == fs) + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x09, 0x1000, 0x1000); + else + rt286_index_update_bits(codec, + NODE_ID_VENDOR_REGISTERS, 0x09, 0x1000, 0x0); + + + return 0; +} + +static int rt286_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + switch (level) { + case SND_SOC_BIAS_PREPARE: + if (SND_SOC_BIAS_STANDBY == codec->dapm.bias_level) + rt286_write(codec, AC_VERB_SET_POWER_STATE, + NODE_ID_AUDIO_FUNCTION_GROUP, AC_PWRST_D0); + break; + + case SND_SOC_BIAS_OFF: + rt286_write(codec, AC_VERB_SET_POWER_STATE, + NODE_ID_AUDIO_FUNCTION_GROUP, AC_PWRST_D3); + break; + + default: + break; + } + codec->dapm.bias_level = level; + + return 0; +} + +static int rt286_probe(struct snd_soc_codec *codec) +{ + hw_configure(codec); + codec->dapm.bias_level = SND_SOC_BIAS_OFF; + + return 0; +} + +static int rt286_remove(struct snd_soc_codec *codec) +{ + rt286_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +#define RT286_STEREO_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) +#define RT286_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) + +struct snd_soc_dai_ops rt286_aif_dai_ops = { + .hw_params = rt286_hw_params, + .set_fmt = rt286_set_dai_fmt, + .set_sysclk = rt286_set_dai_sysclk, + .set_dfs = rt286_set_dai_dfs, +}; + +struct snd_soc_dai_driver rt286_dai[] = { + { + .name = "rt286-aif1", + .id = RT286_AIF1, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT286_STEREO_RATES, + .formats = RT286_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT286_STEREO_RATES, + .formats = RT286_FORMATS, + }, + .ops = &rt286_aif_dai_ops, + }, +}; + +static struct snd_soc_codec_driver soc_codec_dev_rt286 = { + .probe = rt286_probe, + .remove = rt286_remove, + .set_bias_level = rt286_set_bias_level, + .idle_bias_off = true, + .controls = rt286_snd_controls, + .num_controls = ARRAY_SIZE(rt286_snd_controls), + .dapm_widgets = rt286_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt286_dapm_widgets), + .dapm_routes = rt286_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt286_dapm_routes), +}; + +static const struct regmap_config rt286_regmap = { + .reg_bits = 32, + .val_bits = 32, + .reg_write = rt286_hw_write, + .reg_read = rt286_hw_read, + .readable_reg = rt286_readable_register, + .cache_type = REGCACHE_NONE, +}; + +static const struct i2c_device_id rt286_i2c_id[] = { + {"rt5640", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, rt286_i2c_id); + +static struct acpi_device_id rt286_acpi_match[] = { + { "INT33CA", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(acpi, rt286_acpi_match); + +static int rt286_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct rt286_platform_data *pdata = dev_get_platdata(&i2c->dev); + struct rt286_priv *rt286; + struct device *dev = &i2c->dev; + int ret; + + rt286 = devm_kzalloc(&i2c->dev, + sizeof(struct rt286_priv), + GFP_KERNEL); + if (NULL == rt286) + return -ENOMEM; + + rt286->regmap = devm_regmap_init(dev, NULL, i2c, &rt286_regmap); + if (IS_ERR(rt286->regmap)) { + ret = PTR_ERR(rt286->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + i2c_set_clientdata(i2c, rt286); + + if (pdata) + rt286->pdata = *pdata; + + ret = devm_snd_soc_register_codec(&i2c->dev, &soc_codec_dev_rt286, + rt286_dai, ARRAY_SIZE(rt286_dai)); + if (ret < 0) + goto err; + + return 0; +err: + return ret; +} + +static int rt286_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_codec(&i2c->dev); + + return 0; +} + +struct i2c_driver rt286_i2c_driver = { + .driver = { + .name = "rt5640", + .owner = THIS_MODULE, + .acpi_match_table = ACPI_PTR(rt286_acpi_match), + }, + .probe = rt286_i2c_probe, + .remove = rt286_i2c_remove, + .id_table = rt286_i2c_id, +}; + +static int __init rt286_modinit(void) +{ + return i2c_add_driver(&rt286_i2c_driver); +} + +module_init(rt286_modinit); + +static void __exit rt286_modexit(void) +{ + i2c_del_driver(&rt286_i2c_driver); +} + +module_exit(rt286_modexit); + +MODULE_DESCRIPTION("ASoC RT286 driver"); +MODULE_AUTHOR("Bard Liao <bardliao(a)realtek.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/rt286.h b/sound/soc/codecs/rt286.h new file mode 100644 index 0000000..c579c53 --- /dev/null +++ b/sound/soc/codecs/rt286.h @@ -0,0 +1,89 @@ +/* + * rt286.h -- RT286 ALSA SoC audio driver + * + * Copyright 2011 Realtek Microelectronics + * Author: Johnny Hsu <johnnyhsu(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __RT286_H__ +#define __RT286_H__ + +#define NODE_ID_AUDIO_FUNCTION_GROUP 0x01 +#define NODE_ID_DAC_OUT1 0x02 +#define NODE_ID_DAC_OUT2 0x03 +#define NODE_ID_ADC_IN1 0x09 +#define NODE_ID_ADC_IN2 0x08 +#define NODE_ID_MIXER_IN 0x0b +#define NODE_ID_MIXER_OUT1 0x0c +#define NODE_ID_MIXER_OUT2 0x0d +#define NODE_ID_DMIC1 0x12 +#define NODE_ID_DMIC2 0x13 +#define NODE_ID_SPK_OUT 0x14 +#define NODE_ID_MIC1 0x18 +#define NODE_ID_LINE1 0x1a +#define NODE_ID_BEEP 0x1d +#define NODE_ID_SPDIF 0x1e +#define NODE_ID_VENDOR_REGISTERS 0x20 +#define NODE_ID_HP_OUT 0x21 +#define NODE_ID_MIXER_IN1 0x22 +#define NODE_ID_MIXER_IN2 0x23 +#define TOTAL_NODE_ID 0x23 + +#define CONNECTION_INDEX_MIC1 0X0 +#define CONNECTION_INDEX_DMIC 0X5 + +/* SPDIF (0x06) */ +#define RT286_SPDIF_SEL_SFT 0 +#define RT286_SPDIF_SEL_PCM0 0 +#define RT286_SPDIF_SEL_PCM1 1 +#define RT286_SPDIF_SEL_SPOUT 2 +#define RT286_SPDIF_SEL_PP 3 + +/* RECMIX (0x0b) */ +#define RT286_M_REC_BEEP_SFT 0 +#define RT286_M_REC_LINE1_SFT 1 +#define RT286_M_REC_MIC1_SFT 2 +#define RT286_M_REC_I2S_SFT 3 + +/* Front (0x0c) */ +#define RT286_M_FRONT_DAC_SFT 0 +#define RT286_M_FRONT_REC_SFT 1 + +/* SPK-OUT (0x14) */ +#define RT286_SPK_SEL_MASK 0x1 +#define RT286_SPK_SEL_SFT 0 +#define RT286_SPK_SEL_F 0 +#define RT286_SPK_SEL_S 1 + +/* HP-OUT (0x21) */ +#define RT286_HP_SEL_MASK 0x1 +#define RT286_HP_SEL_SFT 0 +#define RT286_HP_SEL_F 0 +#define RT286_HP_SEL_S 1 + +/* ADC (0x22) (0x23) */ +#define RT286_ADC_SEL_MASK 0x7 +#define RT286_ADC_SEL_SFT 0 +#define RT286_ADC_SEL_SURR 0 +#define RT286_ADC_SEL_FRONT 1 +#define RT286_ADC_SEL_DMIC 2 +#define RT286_ADC_SEL_BEEP 4 +#define RT286_ADC_SEL_LINE1 5 +#define RT286_ADC_SEL_I2S 6 +#define RT286_ADC_SEL_MIC1 7 + +#define RT286_SCLK_S_MCLK 0 +#define RT286_SCLK_S_PLL 1 + +enum { + RT286_AIF1, + RT286_AIFS, +}; + +int rt286_jack_detect(struct snd_soc_codec *codec, bool *hp, bool *mic); + +#endif /* __RT286_H__ */ -- 1.8.1.1.439.g50a6b54

3 5